10 Commits

Author SHA1 Message Date
Harald Alvestrand
408cb4bf30 Make SCTPtransport enter "closed" state when DTLStransport does.
Bug: webrtc:11090
Change-Id: I30e0b70387746d6c544ed1818f276569d4258cf4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159888
Reviewed-by: Emad Omara <emadomara@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29810}
2019-11-16 14:56:01 +00:00
Mirko Bonadei
317a1f09ed Use std::make_unique instead of absl::make_unique.
WebRTC is now using C++14 so there is no need to use the Abseil version
of std::make_unique.

This CL has been created with the following steps:

git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt
git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt
git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt

diff --new-line-format="" --unchanged-line-format="" \
  /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \
  uniq > /tmp/only_make_unique.txt
diff --new-line-format="" --unchanged-line-format="" \
  /tmp/only_make_unique.txt /tmp/memory.txt | \
  xargs grep -l "absl/memory" > /tmp/add-memory.txt

git grep -l "\babsl::make_unique\b" | \
  xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g"

git checkout PRESUBMIT.py abseil-in-webrtc.md

cat /tmp/add-memory.txt | \
  xargs sed -i \
  's/#include "absl\/memory\/memory.h"/#include <memory>/g'
git cl format
# Manual fix order of the new inserted #include <memory>

cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \
  xargs sed -i '/#include "absl\/memory\/memory.h"/d'

git ls-files | grep BUILD.gn | \
  xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d'

python tools_webrtc/gn_check_autofix.py \
  -m tryserver.webrtc -b linux_rel

# Repead the gn_check_autofix step for other platforms

git ls-files | grep BUILD.gn | \
  xargs sed -i 's/absl\/memory:memory/absl\/memory/g'
git cl format

Bug: webrtc:10945
Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 15:47:29 +00:00
Bjorn A Mellem
b689af4c99 Changes to enable use of DatagramTransport as a data channel transport.
PeerConnection now has a new setting in RTCConfiguration to enable use of
datagram transport for data channels.  There is also a corresponding field
trial, which has both a kill-switch and a way to change the default value.

PeerConnection's interaction with MediaTransport for data channels has been
refactored to work with DataChannelTransportInterface instead.

Adds a DataChannelState and OnStateChanged() to the DataChannelSink
callbacks.  This allows PeerConnection to listen to the data channel's
state directly, instead of indirectly by monitoring media transport
state.  This is necessary to enable use of non-media-transport (eg.
datagram transport) data channel transports.

For now, PeerConnection watches the state through MediaTransport as well.
This will persist until MediaTransport implements the new callback.

Datagram transport use is negotiated.  As such, an offer that requests to use
datagram transport for data channels may be rejected by the answerer.  If the
offer includes DTLS, the data channels will be negotiated as SCTP/DTLS data
channels with an extra x-opaque parameter for datagram transport.  If the
opaque parameter is rejected (by an answerer without datagram support), the
offerer may fall back to SCTP.

If DTLS is not enabled, there is no viable fallback.  In this case, the data
channels are negotiated as media transport data channels.  If the receiver does
not understand the x-opaque line, it will reject these data channels, and the
offerer's data channels will be closed.

Bug: webrtc:9719
Change-Id: Ic1bf3664c4bcf9d754482df59897f5f72fe68fcc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147702
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28932}
2019-08-21 18:47:58 +00:00
Harald Alvestrand
928e7a3e79 Make ID of datachannel stats not depend on dc.id
The ID of stats was based on the datachannel's "id"
attribute, but that could change - it was -1 before ID
allocation, and a number afterwards.

This CL changes the stats ID to depend on a monotonically
increasing counter for allocated datachannels.

Bug: webrtc:10842
Change-Id: I3e0c5dc07df8a7a502396de06bbedc9f676994a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147642
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28720}
2019-07-31 13:19:08 +00:00
Harald Alvestrand
f3736ed3d8 Datachannel: Use absl::optional for maxRetransmits and maxRetransmitTime.
These parameters are nullable in the JS API.
This allows cleaner handling of "unset" vs "set" in Chrome.

Backwards compatibility note: Behavior should not change, even for users
who set the values explicitly to -1 in the DataChannelInit struct.
Those who try to read back the value will get a compile-time error.

Bug: chromium:854385
Change-Id: Ib488ca5f70bc24ba8b4a3f71b506434c4d2c60b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131381
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27507}
2019-04-09 08:32:43 +00:00
Harald Alvestrand
b9bb3718e3 Revert "Don't crash when a datachannel can't get an ID"
This reverts commit 77c442ca1946924b0acfc9c0ba469ef6a2c3178f.

Reason for revert: Breakage on internal build tests, random breakage elsewhere. Timing issues suspected.

Original change's description:
> Don't crash when a datachannel can't get an ID
> 
> This is exercised by WPT test RTCDataChannel-id.
> 
> Bug: chromium:945256
> Change-Id: I53781dc874134f8c68a49c201848377b93b8858f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128871
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Seth Hampson <shampson@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27300}

TBR=hbos@webrtc.org,hta@webrtc.org,shampson@webrtc.org

Change-Id: Ib7eaf8880e8ce21226b84b3b2283be93acb8dc8b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:945256
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129766
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27302}
2019-03-27 07:23:41 +00:00
Harald Alvestrand
77c442ca19 Don't crash when a datachannel can't get an ID
This is exercised by WPT test RTCDataChannel-id.

Bug: chromium:945256
Change-Id: I53781dc874134f8c68a49c201848377b93b8858f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128871
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27300}
2019-03-26 22:06:24 +00:00
Marina Ciocea
e448a3fb54 Update DataChannel bufferedamount implementation.
Call DataChannelObserver::OnBufferedAmountChange on each successful send.
Previously, the observer would get notified of buffered amount changes only when
queued send data is consumed. Data gets queued only if it cannot be sent right
away. According to the WebRTC standard[1], bufferedamount should be increased
before each sent and decreased after each successful sent. Update implementation
to be standard compliant.

Design doc: http://doc/1lorHBn-GMn5U0T0RQANxrsW0pXhw8XGZM-xZyVUOW90

[1] https://w3c.github.io/webrtc-pc/#dom-datachannel-bufferedamount

Bug: chromium:878682
Change-Id: Ife009d30c4a18dced9a54cf600a445bb1f02561d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123237
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27057}
2019-03-11 13:21:36 +00:00
Steve Anton
10542f21c8 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
Mechanically generated by running this command:

tools_webrtc/do-renames.sh update all-renames.txt && git cl format

Then manually updating:

tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc

Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833
Reviewed-on: https://webrtc-review.googlesource.com/c/115653
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26226}
2019-01-11 17:11:39 +00:00
Steve Anton
1c05765831 (3) Rename files to snake_case: move the files
Mechanically generated with this command:

tools_webrtc/do-rename.sh move all-renames.txt

Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I8b05b6eab9b9d18b29c2199bbea239e9add1e690
Reviewed-on: https://webrtc-review.googlesource.com/c/115481
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26225}
2019-01-11 17:05:20 +00:00